DESPERDICIOS EXISTENTES EN EMPRESAS

APPENDIX A. Client’s Sipura Configuration for PortaSIP

1. First, you need to know the SPA IP address. Via a touchtone telephone attached to the phone port on the SPA, press the star key four times (****). Then type 110# and the IP address will be announced.

2. Run a Web browser application on the same network as the SPA. Open a session in the SPA by typing http://<spa ip address>/admin/advanced.

3. Choose the specific phone port (click on Line 1, Line 2 or another tab).

4. Provide values for the required parameters, which include:

a. in Proxy and Registration:

i. Proxy – PortaSIP® address (or hostname) ii. Register – yes

b. in the Subscriber information part:

i. Display Name – your identification (e.g.

John Doe; this will be seen by the called party)

ii. User ID – SIP account ID

iii. Password – Service password for your SIP account

iv. Use Auth ID – no

5. Submit all the changes and update the SPA configuration.

APPENDIX B. SJLabs Softphone Configuration for PortaSIP

1. First, you need to install the SJPhone on your machine. Following installation, launch the SJPhone software. The following login screen will be displayed.

2. Key in the account ID and password for PortaSIP® and press OK.

The SJPhone display should be similar to the one in the illustration below, showing the account balance in “Ready to call” state. The phone is now ready to be used.

3. Right click on the softphone and press “Login…” to change or make corrections to the account / password.

APPENDIX C. Configuring Windows

Messenger for Use as a SIP User Agent

The following instructions apply to Windows Messenger version 5.0.

1. Start Windows Messenger, and select “Options…” from the

“Tools” menu

2. Check the “My contacts include users of a SIP Communication Service” check box. Enter your “Sign-in name” as shown, in the form username@address, where username is the name of the

appropriate account in PB and address is either the IP address of the PortaSIP® server or its name in DNS. Then click the

“Advanced…” button.

3. Click the “Configure settings” radio button and enter the “Server name of IP address” using either the IP address of the PortaSIP®

server or its name in DNS. Make sure that the “UDP” radio button is selected, then click OK.

4. Sign out and then sign in again. You should see the pop-up dialog below. Fill it in as follows: “Sign-in name” in the form

username@address, where username is the name of the appropriate account in PB and address is either the IP address of the

PortaSIP® server or its name in DNS. Enter the name of the

appropriate PB account as the “User Name” and the appropriate account password as the “Password”, then click OK. You should now see your status change to online.

5. To make a call, click the “Action” item in the main menu, then select “Start Voice Conversation”. Click the “Other” tab, making sure that “Communications Service” is selected in the drop-down Service box, and enter the phone number in the “Enter e-mail address:” field, as shown below. Finally, click “OK” to place a call.

APPENDIX D. Auto-provisioned IP Phones and Adapters

Currently the following IP devices can be auto-provisioned via PortaSwitch®:

 Cisco ATA 186 (firmware versions 2 and 3)

 Sipura 1001

 Sipura 2000

 Sipura 2002

 Sipura 2100

 Sipura 3000

 Linksys PAP2

 Linksys RTP-300

 Linksys/Sipura SPA-2102

 Linksys SPA-942

 Linksys SPA-921

 Linksys SPA-922

 Linksys SPA-3102

 Linksys SPA-941

 Linksys SPA-962

 Linksys WRT54GP2

 GrandStream GXW400x

 GrandStream HT286

 GrandStream HT486

 GrandStream HT488

 GrandStream HT496

 GrandStream HT502

 Thomson TWG850 (only eMTA part)

We are constantly working to extend the list of supported IP devices. If the IP phone you plan to use is not listed here, please contact us – it may already be scheduled for a future release, or we may include it at your request.

APPENDIX E. Configuring Interoperability with DIDX

If your customers plan that calls to DID numbers provided by DIDX will be forwarded to their SIP phones, you need to configure interoperability with DIDX.

Configuration on the PortaSwitch Side

Create a tariff for incoming DID costs and define rates 1. In the Rating section of the Admin-Index page, choose Tariffs.

2. On the Tariff Management page, choose Add.

3. Fill in the Add Tariff form. In the Applied To select menu, choose Vendor. Then clear the Routing check-box, since no routing is actually being done for this vendor, i.e. the vendor will be sending calls to your network.

4. Click Save.

5. Click on the Rates button, then click Add. Enter the rates applied to you by the DID provider.

6. Click Save.

NOTE: The phone prefix for which you are trying to create a rate must already exist in Destinations.

Create a DIDX vendor

This entity is required in order to keep track of your incoming DIDX expenses, and also to provide an adequate level of security on your

network. Since every incoming call to your network must be authorized, you can create an account under the vendor that will be used for such authorization.

1. In the Participants section of the admin interface, choose Vendors.

2. On the Vendor Management page, choose Add.

3. Fill in the Add Vendor form. Please refer to the instructions provided in the Basic SIP Service section for a detailed description.

4. Click Save.

5. Click on the Accounts tab.

6. Click Add to enter a record for the account which will be used by this vendor to send calls to your network.

7. Enter the following values:

 Name – A short name for this account description (visible when associating the account with a connection).

 Login – Enter one of the IP addresses of the DIDX

list, refer to the DIDX Frequently Asked Questions: For Buyers

page.

 Password – Enter cisco.

8. Click the Save button in the toolbar, or the icon on the left end of the row.

9. Click Close in order to return to the Vendors admin page.

Define connections

A connection is the point where calls enter your network via the DIDX gateway.

1. On the Edit Vendor page, press the Connections button.

2. Click Add to add a new connection.

3. Fill in the connection information. Choose VoIP from Vendor as the Type of connection and your PortaSIP® node as the Node, then select the tariff which defines your incoming DIDX costs.

4. In the Vendor Account select menu, choose the account you previously created in the vendor configuration; this account will be used to authorize incoming calls from the vendor.

5. In RTP Proxying, specify this vendor’s capabilities for NAT traversal (if you are not sure whether this vendor can perform NAT traversal, select On Nat).

6. If you would like to use different rate plans for the same service based on where the call comes from, specify the corresponding access code in the Assign Access Code field. This allows the appropriate entry specified in the product’s Services and Rating tab to be selected. For example, the default value for a “VoIP from Vendor” connection (applied to calls to the PortaSIP server arriving from outside your network and terminated to one of your SIP phones) should be set to 'INCOMING'.

7. Description and Capacity are mandatory for all connection types.

8. Click Save.

Configuration on the DIDX Side

1. Sign in to the DIDX members’ area using the membership ID and password you received from DIDX.

2. Click on My Purchased DIDs.

3. View your list of DIDs:

4. Click on the number that you want to point to PortaSwitch®.

5. Select the New SIP option, then enter the SIP address in the format:

<account>@<PortaSIP_proxy> and click Update:

6. Make a test call.